depau/Purism-Calls
1
0
Fork 0
mirror of https://gitlab.gnome.org/GNOME/calls.git synced 2024-06-25 21:29:32 +00:00
Code Issues Releases Wiki Activity

sip: media-pipeline: Take srtp into account when determing pipeline state

Browse source 
If we're using srtp we should also consider the state of srtpenc and srtpdec
elements when determining the state of the whole pipeline.
This commit is contained in:
Evangelos Ribeiro Tzaras 2022-04-07 15:29:39 +02:00
parent bfda8f6a3e
commit 47d4164a09
1 changed files with 11 additions and 11 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

22
plugins/sip/calls-sip-media-pipeline.c
View file

@ -206,22 +206,25 @@ set_state (CallsSipMediaPipeline *self,
static void
check_element_maps (CallsSipMediaPipeline *self)
{
uint all_rtp_elements;
g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
/* TODO take encryption into account */
if (self->element_map_playing == EL_ALL_RTP) {
all_rtp_elements = self->use_srtp ? EL_ALL_SRTP : EL_ALL_RTP;
if (self->element_map_playing == all_rtp_elements) {
g_debug ("All pipeline elements are playing");
set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAYING);
return;
}
if (self->element_map_paused == EL_ALL_RTP) {
if (self->element_map_paused == all_rtp_elements) {
g_debug ("All pipeline elements are paused");
set_state (self, CALLS_MEDIA_PIPELINE_STATE_PAUSED);
return;
}
if (self->element_map_stopped == EL_ALL_RTP) {
if (self->element_map_stopped == all_rtp_elements) {
g_debug ("All pipeline elements are stopped");
set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOPPED);
return;
@ -319,13 +322,10 @@ on_bus_message (GstBus *bus,
else if (message->src == GST_OBJECT (self->rtcp_sink))
element_id = EL_RTCP_SINK;
/* TODO srtp encryption
else if (message->src == GST_OBJECT (self->srtpenc))
element_id = EL_SRTP_ENCODER;
else if (message->src == GST_OBJECT (self->srtpdec))
element_id = EL_SRTP_DECODER;
*/
else if (message->src == GST_OBJECT (self->srtpenc))
element_id = EL_SRTP_ENCODER;
else if (message->src == GST_OBJECT (self->srtpdec))
element_id = EL_SRTP_DECODER;
else if (message->src == GST_OBJECT (self->audio_src))
element_id = EL_AUDIO_SRC;